Tomasz Zieleniewski wrote: > How does asterisk detect the loop. > What are the criteria here. > What do I need to change in the SIP message so > that asterisk will not consider it looped??
> On Nov 23, 2007 4:03 PM, Tomasz Zieleniewski <[EMAIL PROTECTED]> > wrote: > >> hi, >> >> I use asterisk as a gateway which forwards external calls from pstn to >> my internal sip network. >> all sip signaling is passed to sip proxy. >> I also use asterisk as a voicemail server. >> everything works well when calls are passed to asterisk from local >> network. >> but when calls are forwarded from asterisk to sip proxy and then sip >> proxy decides to pass it back to asterisk >> waorking as a voicemail server >> asterisk complains about the loop and returns 482 response. >> Can it be somehow reconfigured?? See http://bugs.digium.com/view.php?id=7403 and look for this code in chan_sip.c: ---cut--- /* Check if this is a loop */ if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->owner->_state != AST_STATE_UP)) { /* This is a call to ourself. Send ourselves an error code and stop processing immediately, as SIP really has no good mechanism for being able to call yourself */ /* If pedantic is on, we need to check the tags. If they're different, this is in fact a forked call through a SIP proxy somewhere. */ transmit_response(p, "482 Loop Detected", req); sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT); return 0; } ---cut--- There's no way to configure the loop detection but you could remove the code to disable loop detection. Grüße, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? -> http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
